The present invention is in the field of audio signal processing and more particularly relates to audio equalizer networks and to hearing assistance systems utilizing such networks.
Audio equalizer networks are used to adjust and/or provide compensation for the frequency response of audio systems. For example, such networks may be used to adjust the frequency response of a loudspeaker system so that the system might have a desired response over specific frequency ranges of interest. Such networks might also be used in a hearing aid system to provide signal level adjustments in specific frequency ranges to compensate for abnormal responses in those ranges of hearing impaired individuals.
Common equalizers consist of a bank of tuned bandpass filters designed for either one octave or 1/3 octave channel spacing. Typical output adjustment ranges are.+-.12 dB for each bandpass channel. First order tuned bandpass filters are often used to avoid the ringing and distortion that result from sharply tuned filters, but such filters are only capable of relatively modest frequency discrimination. Third order 1/3 octave tuned filters offer much better discrimination, but their ringing is often noticeable to the user. Higher order tuned bandpass filters are generally used only to analyze the frequency content of a signal, not to modify it for audio use.
When the outputs of such multiple bandpass tuned filters are combined, the resulting signal typically is characterized by an appreciable amount of amplitude ripple across the composite band. The ripple increases as the adjustment settings increase from the nominal 0 dB setting. 3 to 4 dB of ripple is common in even the best of the tuned filter equalizers at settings of+10 dB. An impulse applied to a tuned filter set with only 1 dB of ripple will produce ringing that is only 20 dB down from the driving signal. This level of ringing is audible to most listeners, and it contributes to the phenomena of sound masking. As a result, the user hears the equalizer instead of the input. In addition to ringing, tuned equalizers also contribute significant amounts of delay and phase distortion. While such equalizers are effective in some applications, they are generally unsuited for use in assisting hearing impaired individuals, who often show amplitude vs. frequency changes exceeding 60 dB at slopes exceeding 20 dB/octave. Tuned filter equalizers cannot compensate for such characteristics.
Moreover, severely hearing impaired individuals have hearing thresholds which may be 90 dB above "normal", but their threshold of pain may not be significantly different from "normal". This phenomena is called recruitment. In most cases, the individual can not permit the use of a level of amplification sufficient to restore a functional ability to hear because of the constant threat of discomfort. The input levels may be carefully controlled in a clinical setting, but real world input levels cannot be controlled. Even normal speech at a distance of 1 meter has a dynamic range which exceeds 20 dB.
Audio engineers face a problem similar to recruitment in preventing high signals from over-loading their systems. Three approaches are generally used to deal with this problem: hard limiting, soft limiting, and compression. Hard limiting prevents excessive signals by simply restricting the maximum signal to a preset level. The part of a signal that exceeds that level is simply cut off. This approach is simple, but it results in a serious amount of distortion. Soft limiting reduces the gain of the amplifier when a signal exceeds a preset maximum level. The signal is not clipped sharply as in hard limiting. Compression controls the output by automatically turning the gain of the amplifier down as the signal increases above a reference level, and up as the signal decreases below a reference level. Compression results in an output signal with a dynamic range that is reduced by some factor (the compression ratio).
Hard and soft limiting and compression have been used in hearing aids for some time. It is known that the human ear compensates for changes in its input level as a function of amplitude and frequency, but the method is not yet understood. While the need for some type of compression mechanism in hearing aids is evident, the implementations to date have not been shown to be consistently beneficial. Single channel compression is not effective because the compression circuitry responds to the highest signal level regardless of its frequency. Loud low frequencies result in the loss of soft high frequencies, for example. In the absence of a signal, a compression amplifier turns the gain up to maximum, resulting in the overamplification of background noise. In spite of these shortcomings in the prior art, it is considered important that some type of compression be used in hearing aid systems in combination with an equalizer network, although the specific format of such compression is not known in the prior art.
Phase or delay compensation as compared to amplitude compensation has not in the past been shown to be significant in assisting hearing impaired individuals. Standard hearing examinations measure only amplitude sensitivity, as recorded in an audiogram. Extreme phase or delay distortion is certainly apparent to most listeners. Most audio equalizer systems and hearing aids concentrate on amplitude compensation and, other than avoiding severe distortion, do little or nothing about phase compensation. When attention is given to phase, one of two approaches is taken. A filter system which has the minimum possible effect on phase is considered desirable, and a filter system which has linear phase characteristics is considered desirable. Neither of these phase characteristics has been shown to be significant in helping hearing impaired individuals. An equalizer system in which both the amplitude and the phase or delay compensation make a significant contribution in improving the hearing and speech discrimination of hearing impaired individuals is not found in the prior art.
Furthermore, individuals whose audiograms are normal or close to normal are considered to have normal hearing even though they may have significant difficulties in the area of speech discrimination. An equalizer system in which the phase or delay compensation alone can make a significant contribution in improving the speech discrimination of such individuals is not found in the prior art.
With specific regard to the use of equalizers in hearing aid systems, audiologists have used audio equalizers to attempt to compensate for hearing loss for many years. Standard one-octave and 1/3 octave audio equalizers consisting of parallel banks of first or third order tuned resonant bandpass filters are often used since they are readily available. A system known as the KSAFA, developed by Dorde Kostic, The Kostic' Methodology for Speech and Language Rehabilitation of Hearing Impaired, University of Wisconsin--Superior, Psycholinguistic Series, VII. 1972 almost 25 years ago, is a system which closely resembles a multi-band audio graphic equalizer. The system uses twenty-seven first order active bandpass filter circuits and corresponding attenuators in parallel to selectively modify the frequency characteristic of an audio source. The Q of each filter can be adjusted to modify the "sharpness" of the frequency response of each band. This approach had several disadvantages: (1) with low Q's, the first order filters did not offer sufficient selectivity, (2) with moderate Q's, the filters were more selective but they did not combine smoothly, (3) with high Q's, the filters hardly combined at all, and they tended to ring, (4) the design did not offer sufficient functional dynamic range to compensate for steep loss slopes, and (5) there was considerable phase distortion, and that distortion varied according to the settings.
More recently, digital signal processing techniques have been used to implement audio equalizers digitally. These digital equalizers are usually based on infinite-duration impulse response (IIR) digital filter realizations, which substantially match the characteristics of the standard resonant tuned filters.
The defining equations for the resonant tuned filters are manipulated using the Z transform to yield equivalent equations which can be easily implemented using digital techniques.
Finite-duration impulse response (FIR) digital filter realizations have also been used, although less often since the IIR approach is simpler. FIR implementations offer several advantages, such as smooth frequency band combining, minimum ringing, and "programmability". However, in connection with hearing aids, there are also several disadvantages:
(1) An equalizer preferably includes independent filter banks for each ear, with center frequencies at approximately 1/4 octave spacing from 200 Hz to 16,000 Hz, resulting in approximately 48 bandpass filters. In addition, some provision for gain control and compression is desirable. These requirements result in a computationally intensive design with the FIR format.
(2) The frequency characteristics need to be modified in a "graphical" manner so that a therapist can identify results of specific changes. This requirement indicates use of complicated Fast Fourier Transforms (FFT) and inverse FFTs, again leading to a high level of circuit or processing complexity.
(3) Time delays between the time that a therapist's lips move and the time that the patient "hears" the speech can be very confusing, much like a movie that has the sound out of synch with the picture. An additional feedback path for many patients is via bone conduction, bypassing the eardrum. In either case, the computational delays of low powered digital signal processing circuits are simply too long. This delay is not a problem in equalizers which are used for normal recording and playback purposes.
(4) The FIR approach requires many delay stages, extremely high clock speeds, and a corresponding high power consumption. "Wearable" aids using this approach are not yet practical.
(5) The linear phase characteristics of FIR filters are not optimum for compensating for hearing loss.
To partially offset some of the disadvantages of the use of FIR digital filters, analog approximations of FIR digital filters might be used. Such analog approximations are disclosed in G. D. Tattersall, "Linear Phase Among Active Filters With Equiripple Passband Responses", IEEE Transactions on Circuits and Systems, Vol. CAS-28, No. 9, Sept. 1981, pp 925-927. The Tattersall article describes a method of realizing a linear phase transversal filter using symmetrical first order all-pass amplifiers and corresponding resistive weights. Each all-pass amplifier is operated only over the limited portion of its frequency response over which its phase characteristics closely approximate those of a pure time delay. This is a marked difference from tuned resonant filters. It is an analog approximation of an FIR filter. When first order all-pass stages are used, there is no frequency aliasing. The Tattersall approach eliminates the computational delays and high power consumption of the digital approach, but it shares the linear phase characteristic of such filters whereby all frequencies are delayed by the same amount. As with its digital counterpart, many delay stages are necessary to achieve good frequency selectivity over the full audio bandwidth. This linear phase limitation makes the Tattersall approach unsuitable for hearing aid applications.
U.S. Pat. No. 4,566,119 (Peters) discloses a variation of the Tattersall approach in the form of analog transversal filters. Those filters use non-symmetrical first order all-pass amplifiers and corresponding resistive weights. They share some of the advantages of FIR filters in that they combine smoothly with minimum ripple and with little ringing. They also display a minimum (rather than linear) phase shift characteristic which is considered desirable in a normal audio equalizer. The equalizers of the '119 patent are capable of modest (.+-.12 dB) gain compensation ranges and have relatively broad bandwidths, characteristics which are desirable for normal audio applications. However, those equalizers are not capable of compensating for the steep loss curves and correspondingly large differences in gain vs. frequency typical of serious hearing loss. And while a minimum phase characteristic may be desirable for standard audio applications, it is not optimum for compensating for hearing loss.
Accordingly, it is an object of the present invention to provide an improved audio equalizer network.
Another object is to provide an improved hearing aid system using such an audio equalizer network.